Abstract: An audio processing system includes at least one first microphone, at least one adaptive filter, and a processor. The at least one first microphone acquires a first audio signal and outputs a first signal based on the first audio signal. The first audio signal includes at least one of a first audio component generated at a first position and a second audio component generated at a second position different from the first position. The first signal is input to the at least one adaptive filter. The at least one adaptive filter outputs a passing signal based on the first signal. The processor, when executing a program stored in a memory, performs: making a determination of which of the first audio component and the second audio component the first audio signal includes more; and controlling a filter coefficient of the adaptive filter based on a result of the determination.

Abstract: An audio processing system includes: a first microphone configured to output a first signal based on a first audio signal; one or more microphones each of which outputs a microphone signal based on an audio signal; one or more adaptive filters configured to respectively receive the microphone signals from the one or more microphones and output passing signals based on the microphone signals; and a processor configured to: determine whether the microphone signal includes uncorrelated noise; control one or more filter coefficients of the one or more adaptive filters; and subtract a subtraction signal based on the passing signals from the first signal. The one or more microphones include a second microphone that outputs a second signal. When determining that the second signal includes the uncorrelated noise, the processor is configured to set a level of the second signal input to the corresponding adaptive filter to zero.

Abstract: The speaker identification system has a voice acquisition unit that acquires voice information of a speaker, and a database management unit that determines whether or not the speaker corresponding to the acquired voice information matches a speaker corresponding to registered voice information in connection with content information on a content, that acquires content information on the content displayed on a device at the time of acquisition of the voice information and stores the acquired content information in connection with the registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information matches the speaker corresponding to the registered voice information, and that stores the acquired voice information in the database as registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information does not match the speaker corresponding to the registered voice information.

Abstract: A hearing aid capable of detecting contact vibration noise from a collected sound signal. The hearing aid is provided with two microphones, a vibration component extracting section that extracts from collected sound signals respectively obtained by the two microphones an uncorrelated component between two collected sound signals as a vibration component for each frequency band. Additionally, a vibration noise identifying section determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting section, an acoustic signal processing section, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver converts the acoustic signal to sound.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A sound processing apparatus which can improve precision of analyzes on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors. The sound processing apparatus is provided with a level signal converter that converts the collected sound signal into a level signal, which indicates an absolute value of the collected sound signal from which phase information is removed. A level signal synthesizer generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors are synthesized, and a detector/identifier carries out analysis on the ambient sounds, based upon the synthesized level signal.

Abstract: A noise cancelling device includes an extracting unit configured to extract a first noise from a signal, the signal being based on an input audio signal, a storing unit configured to store noise characteristic information on a second noise, the second noise remaining after subtracting the extracted first noise from the signal based on the audio signal. And the device further includes a cancelling unit configured to perform cancelling processing for cancelling a noise on the input audio signal based on the first noise and the noise characteristic information on the second noise.

Abstract: The speaker identification system has a voice acquisition unit that acquires voice information of a speaker, and a database management unit that determines whether or not the speaker corresponding to the acquired voice information matches a speaker corresponding to registered voice information in connection with content information on a content, that acquires content information on the content displayed on a device at the time of acquisition of the voice information and stores the acquired content information in connection with the registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information matches the speaker corresponding to the registered voice information, and that stores the acquired voice information in the database as registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information does not match the speaker corresponding to the registered voice information.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A sound processing apparatus (400) is provided with: a directivity synthesis processing unit (410) for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit (440) for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit (451) for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit (452) for adjusting the delay amount such that the difference between the levels becomes smaller.

Abstract: A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(?), based on a power spectrum P1(?) and on a first calculated value obtained by at least multiplying a power spectrum P2(?) by a weight coefficient A2(?). A coefficient update unit (300) updates the weight coefficient A2(?) and a weight coefficient A1(?) so that a second calculated value approximates to the power spectrum P1(?). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(?) and the estimated target sound power spectrum Ps(?) by the weight coefficient A2(?) and the weight coefficient A1(?), respectively.

Abstract: The present invention provides a hearing aid capable of detecting contact vibration noise from a collected sound signal. A hearing aid (100) is provided with two microphones (110-1, 110-2), a vibration component extracting unit (120) which extracts from collected sound signals respectively obtained by the two microphones (110-1, 110-2) an uncorrelated component between two collected sound signals as a vibration component for each frequency band, a vibration noise identifying unit (130) which determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting unit (120), an acoustic signal processing unit (140) which, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver (150) which converts the acoustic signal to sound.

Abstract: A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(?), based on a power spectrum P1(?) and on a first calculated value obtained by at least multiplying a power spectrum P2(?) by a weight coefficient A2(?). A coefficient update unit (300) updates the weight coefficient A2(?) and a weight coefficient A1(?) so that a second calculated value approximates to the power spectrum P1(?). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(?) and the estimated target sound power spectrum Ps(?) by the weight coefficient A2(?) and the weight coefficient A1(?), respectively.

Abstract: A sound processing apparatus (100), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110-1 and second sound collector 110-2), and the sound processing apparatus (100) is provided with a level signal conversion section (first level signal conversion section 130-1, second level signal conversion section 130-2) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section (140) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110-1 and second sound collector 110-2) are synthesized, and a detecting and identifying section (160) that carries out analysis on the ambient sounds based upon the synthesized level signal.

Abstract: It is an object of the present invention to provide an audio signal encoding method, an audio signal decoding method, a transmitter, a receiver, and a wireless microphone system which can compress an audio signal at a relatively high compression ratio at a relatively high quality with a relatively low delay. The compression encoder 4 of the transmitter includes an audio signal dividing filter bank 4a for dividing the audio signal into four sub-band signals, sampling each of the sub-band signals at a down-sampling rate depending on the number of the sub-band signals, and producing the sub-band signals sampled at the down-sampling rate, LD-CELP encoders 20a to 20d for encoding the sub-band signals on the basis of LD-CELP algorithm, and a multiplexer 4c for producing a multiplexed data stream with the encoded sub-band signals.

Abstract: An audio encoder converts an input sound signal into a plurality of compressed frame data pieces in an sound signal compression coder, determines the importance of each bit in a classification unit of a transmission line coder based on the decoding quality in the presence of a transmission error, and classifies the bits into a plurality of classes. The audio encoder selects one of the three types of processing including convolution coding and addition of CRC check codes, convolution coding only, and no coding, in descending order of importance in the presence of a transmission error for each class. Then, the audio encoder adds preamble information and a synchronization signal in a multiplexer to generate a bit stream. It becomes possible to suppress degradation of a decoded sound signal without additional redundant bits.

Abstract: An audio signal coding device, an audio signal decoding device and a method to improve audio quality. The audio signal coding device and method include a quantizer that quantizes a given signal according to a number of assigned bits in order to generate a codeword. The coding device includes an extractor that extracts core bits from the generated codeword. The coding device also includes a determiner that determines an optimal value of the number of assigned bits based on an energy level corresponding to the extracted core bits. The audio signal decoding device and method include a dequantizer that dequantizes a given codeword according to the number of assigned bits to generate a decoded signal. The decoding device includes an extractor that extracts core bits from the given codeword. The decoding device also includes a determiner that determines an optimal value of the number of assigned bits used in the dequantizer.

Abstract: An apparatus having a band-separating filter bank for separating a digital signal into a plurality of sub-band signals, to be processed or transmitted, and a band-combining filter bank for subsequently combining the resultant sub-band signals into a single digital signal, wherein each of the band-separating filter bank and band-combining filter bank incorporates a FIR low pass filter having an asymmetric impulse response, as the prototype filter of the filter bank. A significant reduction can thereby be achieved in the amount of overall group delay that results from the processing performed by these filter banks.

Abstract: An apparatus having a band-separating filter bank for separating a digital signal into a plurality of sub-band signals, to be processed or transmitted, and a band-combining filter bank for subsequently combining the resultant sub-band signals into a single digital signal, wherein each of the band-separating filter bank and band-combining filter bank incorporates a FIR low pass filter having an asymmetric impulse response, as the prototype filter of the filter bank. A significant reduction can thereby be achieved in the amount of overall group delay that results from the processing performed by these filter banks.